Hello Kevin,
> If your keys have passphrases, then you must run the 'init keys' CLI
> command to enter the passphrase before the keys can be loaded.
>
> It's much simpler to just not use passphrases and instead protect your
> private key every other way that you can :-)
Yeah I rebuilt my keys w
On Mon, 31 Oct 2005, Folkert van Heusden wrote:
> In sip.conf I have this:
> [1000]
> type=peer
> host=dynamic
> defaultip=192.168.62.100
> dtmfmode=rfc2833
> mailbox=
> context=dialoutcont
> callerid="Folkert van Heusden" <[EMAIL PROTECTED]>
You've defined a peer - which is for calls TO th
The second beta of Asterisk 1.2.0 has been released! It is available from
the ftp.digium.com FTP servers, as well as the Digium CVS servers (under
the 'v1-2-0-beta2' tag).
This beta includes a large number of improvements over beta1, including:
* Many, many bug fixes
* Documentation and samp
That patch may or may not work. It just depends on which day you grabbed
the asterisk code. This is why I stopped reponding to requests about the
makefile patch failing. It is simply impractical to offer a working
solution for everyone. The Makefile has changed frequently, and often in
ways tha
Asterisk version
mm*CLI> show version
Asterisk 1.0.9 built by [EMAIL PROTECTED] on a i686 running Linux
Zaptel, I am not sure but If you have built from CVS, the version info
should be in the .version file in the src directory
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mai
On Mon, Oct 31, 2005 at 10:48:44AM -0800, Bart Fisher wrote:
> Is there a command line for discovery of Asterisk and Zaptel Versions?
I'm not aware of any way to tell the version of the zaptel kernel
module. But then again, on my system there is the indirect way of 'dpkg
-l zaptel-modules\*'
--
On Mon, Oct 31, 2005 at 04:14:39PM -0800, Andy Kuo wrote:
> Hi all,
> I'm trying to install spandsp. I followed the instructions on
> http://www.soft-switch.org/installing-spandsp.html, and when I applied the
> patch, I got the following errors:
> [EMAIL PROTECTED] apps]# patch < apps_makefile.pa
I have RTFM, Googled and experimented on this issue, but it's
still unclear.
In the sip.conf file, what's the difference between:
secret=welcome
and
password=welcome
The O'Reilly book example uses the first version, but I can't
get no registration with it. The only way my SIP phones ar
OTR Comm wrote:
I did give the key a passphrase, could this be the problem? Do I have
to put the passphrase in some config file?
If your keys have passphrases, then you must run the 'init keys' CLI
command to enter the passphrase before the keys can be loaded.
It's much simpler to just not
Remco,
Did you figure out what was happening, did you get it working?
Regards,
PolAus
_
SEEK: Over 80,000 jobs across all industries at Australia's #1 job site.
http://ninemsn.seek.com.au?hotmail
_
http://zarzamora.com.mx/asterisk/48
I know it´s a tutorial for E1/R2 but it has a new patch that works with CVH-HEAD and spandsp. Try it out.2005/10/31, Andy Kuo <[EMAIL PROTECTED]
>:Hi all,
I'm trying to install spandsp. I followed the instructions on http://www.soft-switch.org/installing-span
Others have also observed consistent echo problems with the
mec2 and kb1 cancellers when installed against a channel bank. Likely this has
been addressed by the patches incorporated into the new mg2 derivative which
attempt to deal more correctly with really wide ranging signal levels. It wou
One trusts that the vendor or service provider knows their
system best... however, to answer your question; yes, it's plausible that the
choice of codecs _could_ interfere with proper echo cancellation. Scenarios that
spring to mind include an improperly implemented codec that is causing the
Hi,
I've got a strange problem here: I am using Asterisk 1.0.7 (the Debian
Testing version asterisk-1.0.7.dfsg.1-2). The dtmfmode is set to rfc2833
and I'm making a call to Asterisk via SIP. If I press star in this call,
it is recognized as a pound and the star is not available. The physical
poun
Hello all,
I am having a terrible time with RSA registry. I have genetrated a key
for 'pbx1too' with astgenkey and put it in /var/lib/asterisk/keys/, but
when I reload iax2 I get an error:
authenticate: Unable to find private key 'pbx1too'
Where else would asterisk be looking for the key?
I al
I'm having the same problem. Can anyone help? I'm on the CVS 1.1 version.On 10/31/05, Andy Kuo <[EMAIL PROTECTED]
> wrote:Hi all,
I'm trying to install spandsp. I followed the instructions on http://www.soft-switch.org/installing-spandsp.html
, and when I applied the patch, I got the following e
Sorry guys, I forgot to add; It works fine if
set-up in the features.conf file, but not when the dedicated transfer button on
the phone is used.I know it uses the refer method for this, but this
just creates zombies according to the * console
output
__
Sorry guys, I forgot to add;
It works fine if set-up in the features.conf file,
but not when the dedicated transfer button on the phone is used.
I know it uses the refer method for this, but this
just creates zombies according to the * console
output
__
I wonder if anyone can help. I've built an asterisk instance against
the latest 1.1 CVS version. Can anyone help me out with a makefile for
rx_fax?
I'm following:
http://soft-switch.org/installing-spandsp.html
I have built spandsp OK, but get errors when I'm applying the patch.
Unfortunately I'm
Thanks for the suggestion. Any idea where I can buy them?
massimo wrote:
Hi Stephen, I don't think you can use fritz card
to connect to a Siemens pbx.
You have to use a card that works in NT mode for
exemple a more cheap compatible Bristuff card.
Refer to this page:
http://www.voip-info.org/wik
Ronald Hartmann wrote:
NOTE the discrepancy with musiconhold versus musicclass not sure if I am
reading this correctly but looks like we need to use both musicclass and
musiconhold
This was just corrected in CVS HEAD this evening.
___
--Bandwidth and
Hi, I’m having a problem with my IAX2 VOIP service
provider in that a large number of calls now exhibit varying echo at least 1 in
3 calls outgoing and almost 50/50 incoming.
They have just replied saying it is codec related and for me
to change codecs.
Is this for real? Can echo be
Good Day list,
I am having a bit of an issue as it relates to the musiconhold
settings in the Version 1.09 of Asterisk
Problem I am unable to set different music classes for different
extensions.
1) (default) I would like to be able to set generic music on hold for
the company, extensio
2005/10/31, Walter Willis <[EMAIL PROTECTED]>:
Tengo un asterisk en al oficina de un cliente , que quiere hacer
llamadas ilimitadas a estados unidos; las llamadas tienen que ser al
mismo tiempo.
alguien ofrecio una conexion iax2 para los 4 usuarios.
no tienes algo mas atractivo se supone que func
Greetings!
I am running a small callcenter with 10
analog lines, aprox. 15 agents and using Asterisk 1.2beta1. We have 10
sipura 3000s connected to the PSTN and a few linksys PAP2s.
The ports connected to phones are configured as
SIP/200s and SIP/300s and the ones connected to the PSTN as
Hi all,
I'm trying to install spandsp. I followed the instructions on http://www.soft-switch.org/installing-spandsp.html, and when I applied the patch, I got the following errors:
[EMAIL PROTECTED] apps]# patch < apps_makefile.patchpatching file MakefileHunk #1 FAILED at 55.Hunk #2 FAILED at
Hello all,
I am having D-channels problems.
I have setup Asterisk (CVS-HEAD-10/31/05-16:01) on RH Fedora Core
4(2.6.13-1.1532_FC4). I also have 2 TE110XP cards set to T1 (only one
with a T1 connect currently). I had my telco provider rebuild the T1 on
a new card, and they say that they can see
Version 0.7 of HooDaHek, the CallerID Notification System for Asterisk,
has been released. This is the last version with this database schema;
the next will be breaking things into many more fields of information,
including a lot of custom actions for calls (sending calls to VM by
CLID, ignori
estoy en peru y tengo un sppedy 900 con 1200 de subida y 256 de bajada2005/10/31, Carlos Alperin <[EMAIL PROTECTED]>:
Si, nosotros podemos
darte ese servicio. Necesitas DID en que ciudad?
Podemos conectarte via
IAX2 o SIP.
Carlos Alperin
Senior System Engineer
Seneca Com
Si, nosotros podemos
darte ese servicio. Necesitas DID en que ciudad?
Podemos conectarte via
IAX2 o SIP.
Carlos Alperin
Senior System Engineer
Seneca Communications, LLC
[EMAIL PROTECTED]
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Walter Willis
your conecct asterisk to service http://connect.voicepulse.com/
how to ???
diagram
client1-> asteriskprovider iax (four lines) (for call USA unlimited phones)
client2--/
client3-/
client4/
exist the service???
2005/10/31, Rusty Dekema <[EMAIL PROTECTED]>:
No hablo español muy bueno
On Oct 30, 2005, at 3:08 PM, Eric Bishop wrote:
I am running CVS HEAD. How can I tell which software echo canceller I
am using?
Look in zconfig.h within the zaptel package. It will tell you.
Matthew Fredrickson
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Just a point of reference for others, we had a lot of echo on an Adtran Channel Bank connected to a T100P.
The end poitns were all Polycom Ip500's on a local lan connecting to *.
We couldn't solve the echo with any software settings and installed a
Tellabs Echo Can shelf 255D i believe with 64ms e
I am looking to buy wither the 1U or desktop E1 echo canceller from
Orion. Has anyone had any experiences either good or bad with these
units?
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Asterisk-Users@lists.
On Oct 31, 2005, at 3:12 PM, [EMAIL PROTECTED] wrote:
Yes, using and analog with ATA is an option, but one of the
requirements is
to avoid eletric power cabling, and there is an explicit request
for Power
Over Ethernet phones (which adds another not-so-common feature)...
so a
native VoIP p
On Oct 30, 2005, at 2:06 PM, Wayne wrote:
Stefan Gofferje wrote:
Hi folks,
whoever owns a Cisco phone and is unhappy about slow firmware,
incomplete XML support etc... should really have a look at Sergio
Chersovani's rewrite of chan-sccp!
Ok - I'll give it a go :) - Just one problem...
Hello Bart ,
On Mon, 31 Oct 2005, Bart Fisher wrote:
> Thanks, but what I was really hoping for was something that could be used in a
> script to report current revisions... me sad
> Bart
> - Original Message - From: "C F" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - No
Hello Jorge & All ,
On Mon, 31 Oct 2005, Jorge Merlino wrote:
> There is the -T option when running the CLI but I think it only works in 1.2
>
> Regards
> Jorge
>
> El Lun 31 Oct 2005 12:30, [EMAIL PROTECTED] escribió:
> > Hello!
> >
> > Lately, I've been keeping a close eye on an
Thanks, but what I was really hoping for was something that could be used in
a script to report current revisions... me sad
Bart
- Original Message -
From: "C F" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, October 31, 2005 12:33 P
After reading through the rfc http://www.ietf.org/rfc/rfc2976 and cisco
site
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftinfo.htm
And googling a bit I can not find anything that look out of place, I
notices a few formating difference on examples I hav
Hi,
I'm running asterisk 1.0.9 with an FXO card. People can call me on my pstn line
and that gets transferred to my laptop (on 192.168.62.100). That all runs fine.
If, though, I want to dial out (a pstn line) I always get a "call rejected: 404
not found" error (in sjphone) or a plain "not found"
Excellent guys. This has to be the quickest responce I have ever had
on this list!
Scott.
On 10/31/05, Peer Oliver Schmidt <[EMAIL PROTECTED]> wrote:
> Scott schrieb:
> > Is it possible to schedule dymanic queues?
> >
> > Currently I have a queue that has dynamic members of which I would
> > lik
Scott wrote:
Currently I have a queue that has dynamic members of which I would
like to set a schedule for. From say 8am to 5pm the queue would ring
the phones of queue members but after 5pm the caller get's VM.
Of course it is... don't send the call into the queue if the call
arrives outsi
Scott wrote:
Is it possible to schedule dymanic queues?
Currently I have a queue that has dynamic members of which I would
like to set a schedule for. From say 8am to 5pm the queue would ring
the phones of queue members but after 5pm the caller get's VM.
Is this possible?
Thanks.
Scott.
__
Scott wrote:
Is it possible to schedule dymanic queues?
Currently I have a queue that has dynamic members of which I would
like to set a schedule for. From say 8am to 5pm the queue would ring
the phones of queue members but after 5pm the caller get's VM.
Is this possible?
Thanks.
Scott.
__
Scott schrieb:
Is it possible to schedule dymanic queues?
Currently I have a queue that has dynamic members of which I would
like to set a schedule for. From say 8am to 5pm the queue would ring
the phones of queue members but after 5pm the caller get's VM.
Do it in the dialplan by branching
Scott wrote:
Is it possible to schedule dymanic queues?
Currently I have a queue that has dynamic members of which I would
like to set a schedule for. From say 8am to 5pm the queue would ring
the phones of queue members but after 5pm the caller get's VM.
Is this possible?
Yes, this is simpl
Yeah, show versions in the CLI will give you the version of your asterisk build
Also you can do the following in the CLI:
show version files
where is a valid file name.
As always in Linux you can press TAB to get a list of available
commands in the CLI, for example you can type:
show version file
Is it possible to schedule dymanic queues?
Currently I have a queue that has dynamic members of which I would
like to set a schedule for. From say 8am to 5pm the queue would ring
the phones of queue members but after 5pm the caller get's VM.
Is this possible?
Thanks.
Scott.
__
Hi;
what's card do u use 5 v or 3.3 v?
u can find motherboard in :
http://www.digium.com/index.php?menu=compatibility
http://64.233.183.104/search?q=cache:UqnvELBs-AUJ:www.voip-info.org/wiki/view/Asterisk%2Bhardware+motherboard+for+digium+card&hl=fr
I hope that can help u
2005/10/31, Ken Dresdel
Yes, using and analog with ATA is an option, but one of the requirements is
to avoid eletric power cabling, and there is an explicit request for Power
Over Ethernet phones (which adds another not-so-common feature)... so a
native VoIP phone would be welcome.
Francesco Pellegrini
__
On 10/31/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
> Is there a way to add timestamps to each line in the console so you know
> exactly how long a call took? Or is there another way of telling directly
> within the console?
>
I must say its something I would really like to see on the con
There is the -T option when running the CLI but I think it only works in 1.2
Regards
Jorge
El Lun 31 Oct 2005 12:30, [EMAIL PROTECTED] escribió:
> Hello!
>
> Lately, I've been keeping a close eye on an Asterisk box by staying logged
> into the console for long periods of time. However, i
On 10/31/05 23:51 Fabio Montemaggiore said the following:
Why Asterisk show this message?
WARNING[14792]: chan_iax2.c:9355 load_module: Unable
to open IAX timing interface: No such device
it's just a warning. without a timing device, you couldnt use IAX2
trunking, which would greatly reduce
Rich Adamson wrote:
Assuming you are using the TDM card, there is no code in asterisk to detect
whether a pstn line is connected/disconnected, nor does it listen for dialtone
before dialing.
And for some reason this isn't considered a SEVERE defect?
If the battery on the line disappears,
Hi,
I wonder if anyone can help. I've built an asterisk instance against
the latest 1.1 CVS version. Can anyone help me out with a makefile for
rx_fax?
I'm following:
http://soft-switch.org/installing-spandsp.html
I have built spandsp OK, but get errors when I'm applying the patch.
Unfortunately
CVS HEAD as of two days ago.
We have 50 agents (All SIP, with inbound/outbound via ISDN32 using a
TE405P with revision 2 firmware), logging in via agentcallback. At the
start of every day I restart * (service restart) At the end of today
(and most other days) we have the following problems:
Hello everyone,
We are experimenting a really bad sound quality with a Digium card and the
technical support from Digium found out that we have a motherboard
incompatible with the card.
If that can help anyone, here are 2 motherboards that we have tested with
very bad "zttest" results :
Intel
[EMAIL PROTECTED] wrote:
Does anyone know where the official release of Asterisk 1.2 is? Do we have
a time-frame of when this version will be released and how much longer it
will be in BETA.
'Where' it is? It's in the future :-)
There is no time frame, as has already been discussed on this lis
On 10/31/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Does anyone know where the official release of Asterisk 1.2 is? Do we have
> a time-frame of when this version will be released and how much longer it
> will be in BETA.
This question was answered *yesterday* by Mr. Olle E. Johansson:
An
Is there a command line for discovery of
Asterisk and Zaptel Versions?
Bart
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http://lists.digium.com/mailman/listinfo/asterisk-us
On 10/31/05, John Fraser <[EMAIL PROTECTED]> wrote:
> cannot overwrite directory `/var/www/asterisk2/htdocs/./A2Billing_UI
> cannot remove `/var/www/asterisk2/htdocs/./A2Billing_UI': Is a directory
:-/
Try with cp -rf
>
> Ill just start from scratch.
> could you put a version number and date on th
tengo un asterisk, alguien conoce algun proveedor que brinde el
sistema de linkar mi asterisk a su servicio para tener tarifa plana a
eeuu.
para llamar por 4 conexiones al miamo tiempo desde mi asterisk?
me parece haber visto que se configuraba con una troncal iax2
2005/10/31, [EMAIL PROTECTED
Is there a resolution to this problem. It was posted a few weeks back.
But just chiming in again to see if someone has had any luck:
Problem:
Incoming call to a Sipura 2000, 1000, 3000 ATA.
I use the SetCallerID(name)="blah blah blah"
SetCAllerId(number)="1234567890"
However, On the handset,
In his Outgoing context, should it not be 9|1NXXNXX, to strip the 9 from
being sent to the provider?
Kevin
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: October 31, 2005 7:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Sub
Does anyone know where the official release of Asterisk 1.2 is? Do we have
a time-frame of when this version will be released and how much longer it
will be in BETA.
Thanks.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
Hello,
I am using Cisco 7940/7960 phones and can not get the calling name to
display on incoming calls. The names and numbers do display in the
Missed Calls and Received Calls menus, but not on incoming. The caller
id number displays fine on incoming, just not the name. Anybody know
what might
you should use an analog made for ships with a ATA.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Monday, October 31, 2005 7:13 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Rugged VoIP phones for use with asteri
I have a codeless language module for BBEdit, if anyone's interested
- it's not complete yet (I'm adding to it as I go along), but I will
post it to the wiki, if I could just figure out where it should
go...
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouve
Hi,
Thanks for the clarification. I had seen that the two options
existed, but the docs for the dial() command didn't state the
difference.
On Sun, Oct 30, 2005 at 08:23:32PM -0500, David Bandel wrote:
> On 10/30/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > Hi All,
> >
> > Recently got
Yes, or this for example:
[macro-rhangup]
exten => s,1,NoOp(DIALSTATUS=${DIALSTATUS})
exten => s,n,NoOp(TIME=${DATETIME})
exten => s,n,Hangup
I also output the date and time prior to dialing out.
MARK.
Sherwood McGowan wrote:
You could always just add some
exten = NUM,PRIO,VERB
Darren,
An example how to call that callback.agi script? The script iself does
not have usage info.
Thanks
Anthony
Darren Wiebe wrote:
Hello. You should not need any special hardware for callback. You will
(obviously) need card to connect your box to the pstn. Do you have
something s
Hi,
I seek solution for hotel management and billing solution. but I do not
know which to choose between Astbill or Asterbill ? if you have council.
Thx
David
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cannot overwrite directory `/var/www/asterisk2/htdocs/./A2Billing_UI
cannot remove `/var/www/asterisk2/htdocs/./A2Billing_UI': Is a directory
Ill just start from scratch.
could you put a version number and date on the web page please.
Might save me and others some trouble
On Mon, 31 Oct 2005 10:3
In article <[EMAIL PROTECTED]>,
Chris Bagnall <[EMAIL PROTECTED]> wrote:
>
> exten => _44N.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
> (N should be the same as [1-9] I think)
N is [2-9], Z is [1-9]
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PR
I don't set the mailserver.
What can I do?
I use Debian
Thanks
___
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http://mail.yahoo.it
___
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In article <[EMAIL PROTECTED]>,
James Steven <[EMAIL PROTECTED]> wrote:
> Hi
>
> I have inserted the lines you suggested but Asterisk keeps the 0 when
> dialling with all alternatives. Also, I am unsure what the phrase
> "${EXTEN::2}${EXTEN:3}" does? Could you explain this abit?
>
> My extensi
> I have inserted the lines you suggested but Asterisk keeps
> the 0 when dialling with all alternatives. Also, I am unsure
> what the phrase "${EXTEN::2}${EXTEN:3}" does? Could you
> explain this abit?
The syntax is {EXTEN:initial offset:length}
So EXTEN:3 chops off the first three digits a
did u set the mailserver?
Bruno.
Fabio Montemaggiore wrote:
> I don't receveid e-mail with voicemail.
> When I dial 2 with telephone, Asterisk record message
> but don't send a e-mail at the mailbox. Why?
> I have configuration this file.
>
>
>
> In the voicemail.conf
> [general]
> attach=yes
> fo
Why Asterisk show this message?
WARNING[14792]: chan_iax2.c:9355 load_module: Unable
to open IAX timing interface: No such device
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__
"Olle E. Johansson" <[EMAIL PROTECTED]> wrote:
>Adam Moffett wrote:
>> does anyone know when 1.2 will no longer be beta?
>>
>The quick answer is: When it's ready for release.
>
>Open Source software doesn't really follow a set agenda.
I don't think that is an accurate statement. It is certainl
I don't receveid e-mail with voicemail.
When I dial 2 with telephone, Asterisk record message
but don't send a e-mail at the mailbox. Why?
I have configuration this file.
In the voicemail.conf
[general]
attach=yes
format=wav
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
sendvoicemai
No you won't !!! You just have to copy again the AGI & Web Interface
You don't have to change anything in your configuration files or
in your Database. It should be really fast to do!
FYI -> areski.net/a2billing
## (ADMIN UI V1.0.3 - CUST UI V1.0.1 - AGI V1.0.1 - DB V1.0.0)
this will keep you infor
Hi
I have inserted the lines you suggested but Asterisk keeps the 0 when
dialling with all alternatives. Also, I am unsure what the phrase
"${EXTEN::2}${EXTEN:3}" does? Could you explain this abit?
My extensions.conf is:
[default]
exten => _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
Hi,
Is it possible to dynamically add contexts to the dial plan in any way?
Extensions can be added from the console and therefore also from MAPI but
their doesn't appear to be anyway to add a new context apart from reloading
the configuration files.
The reason I ask is my dialplan is getting qu
Does anyone have an example of a lucent
TNT h323 config to work with asterisk ?
I'd like to use sip but it's not supported in the
TAOS we have, if anyone has TAOS 10.x or later
that would be awsome as well, we have the examples
for a sip config.
thx
- Armand
__
Hey everybody.
I have an Adit 600 that I'm not able to get working properly with
Groundstart. The Adit (BootCode 1.23), 1 TDM (Version 6.1.2) and 1 FXO
card (Version 1.12).
The Adit is setup: ESF,B8ZS.
1st port is set as signal gs type voice.
2-8 is setup signal ls type voice.
The FXO c
Is there anyone who knows where to find rugged IP phones?
Rugged in this case means that need to be installed on a ship's deck, so it
must be water resistant, anyway compliant with IP 65 specification
(protected against dust and jets of water).
Regards
++
| Fran
Brent Torrenga wrote:
So within the /var/lib/sounds/voicemail structure are the greeting files
recorded by the person at each extension (busy.wav, greet.wav). If I need to
get rid of the customized recording, it is trivial to simply delete both of
those files. At that point, if a call goes to voi
Hi,
would I have to go through the entire installation again?
Thanks
John Fraser
On Mon, 31 Oct 2005 09:46:44 -0500, Areski K wrote
> Please download the last release (http://areski.net/a2billing/),
> I corrected some bugs and this was one of them.
> Rgds, Areski
>
> On 10/31/05, John Fraser <
I added versioning information (version.release) in the application
(ie agi ./a2billing --version).
It will be more easy to know which version, release you downloaded and
check if a new one is available.
# Last release have an ACL user support & also advanced filter to
select the cards.
Rgds, A.
Why'd your mailing software fake your From: to be my email address?
On Monday 31 October 2005 09:35, [EMAIL PROTECTED] wrote:
> I started with a Voicetronix Openswitch12, which has, even in its driver
> code, no support for anything resembling kewlstart. After figuring out
> that the MICS seemed
I have a beta1 gateway with a 4 port card in PRI mode, the gateway sends
DTMF via sip info packets to another beta1 box. The peer is set to
receive info. What I get is a click sound and a very very short tone.
Sound like to me that I get the first part of the tone before it is
captured and put
Joining two conferences together over a LAN should be possible, at least
theoretically. I am not sure how the performance would be over a WAN or the
public internet. I am currently working on joining two meetme conferences
together using IAX2 trunking and will post my results after the trial.
An
So within the /var/lib/sounds/voicemail structure are the greeting files
recorded by the person at each extension (busy.wav, greet.wav). If I need to
get rid of the customized recording, it is trivial to simply delete both of
those files. At that point, if a call goes to voicemail, then Allison wil
Please download the last release (http://areski.net/a2billing/),
I corrected some bugs and this was one of them.
Rgds, Areski
On 10/31/05, John Fraser <[EMAIL PROTECTED]> wrote:
> I am getting the following error when I click on "create new ratecard"
>
> Fatal error: Cannot redeclare display_minut
You could always just add some
exten =
NUM,PRIO,VERBOSE(LEVEL|${DATETIME} -- THIS EVENT HAPPENED)
type commands in your dialplan to force output of the date
time, and you can even reduce the amount of verbosity to the CLI by using it
liberally to signify events, so you don't have to
O'reilly had a book out before the docs team wrote theirs.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - -- - -- -- - --
"Leif Mad
Hi Andrew,
Thanks for responding. Yes, I noticed that some code *seems* to support
this.
I started with a Voicetronix Openswitch12, which has, even in its driver
code, no support for anything resembling kewlstart. After figuring out that
the MICS seemed to respond to opening the circuit for a s
Rich Adamson wrote:
Just update cvs-head again at 7:45pm CST. Seems the issue still exists.
Any thoughts on me opening a bug tracker item on this?
Always a good idea (and cheap too!)
___
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Asteri
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